Adaptive bitrate management for streaming media over packet networks

ABSTRACT

A method including receiving a receiver report from a terminal; estimating one or more network conditions of a media network based at least in part on the receiver report; determining an optimal session bitrate based on the estimated one or more network conditions; and providing media data to the terminal based on the optimal session bitrate.

CROSS REFERENCE TO RELATED PATENTS

This application is a continuation of U.S. application Ser. No.12/170,347, filed Jul. 9, 2008 now U.S. Pat. No. 7,987,285, “AdaptiveBitrate Management For Streaming Media Over Packet Networks,” whichclaims the benefit of U.S. Provisional Application No. 60/948,917, filedJul. 10, 2007, “Adaptive Bitrate Management For Streaming Media OverPacket Networks,” both of which are incorporated herein by reference.

BACKGROUND INFORMATION

Rate control is essential for media streaming over packet networks. Thechallenge in delivering bandwidth-intensive content like multimedia overcapacity-limited, shared links is to quickly respond to changes innetwork conditions by adjusting the bitrate and the media encodingscheme to optimize the viewing and listening experience of the user. Inparticular, when transferring a fixed bitrate over a connection thatcannot provide the necessary throughput, several undesirable effectsarise. For example, a network buffer may overflow resulting in packetloss causing garbled video or audio playback, or a media player buffermay underflow resulting in playback stall. Standard bodies haverecommended protocols to address these issues. Internet Engineering TaskForce (IETF), in RFC 3550, specifies RTCP as the fundamental buildingblock to implement bit rate/packet rate control in streaming media.Several extensions to RTCP, suited for high capacity networks, followthis original recommendation.

Even with these recommended protocols, delivering a multimedia sessionover wireless networks can be particularly challenging, due in part tothe following:

-   -   Sudden Adjustment of nominal transmission rate: Due to        interference, fading, etc, 3+G networks negotiate physical layer        parameters on the fly. Nominal transmission bitrates can change        by a factor of 10;    -   Packet Loss: caused by either link transmission errors or by        network congestion;    -   Reduction of Effective bandwidth: The wireless link is a shared        resource at Layer 2, with MAC (Media Access Control) mechanism        and scheduling. This means that an increased load presented by        other wireless terminals in the same sector can reduce the        effective bandwidth or capacity that a terminal will see; and    -   Limited Capacity: Available capacity is typically a fraction to        that obtained in traditional wireline internet access        technologies, where currently capacity is not an issue. Fixed        internet media sessions can typically offer to the network loads        between 250 and 400 kbps. Despite the fact that current 3G        cellular networks can sustain throughputs of 500 kbps and above,        the total bitrate budget for a wireless multimedia session is        typically kept under 150 kbps to ensure scalability.

For wireless mobile devices, providing a good experience in streamingmedia sessions is particularly difficult, due to

-   -   Infrequent and incomplete network state information. The typical        wireless media player support RTCP receiver report as defined in        RFC 3550, and the report generation frequency is fixed. As a        result, the network state information obtained at the sender end        is limited and sporadic. In its Packet Streaming Service        specification, 3GPP recommends several extensions to the basic        IETF RTCP Receiver Report (i.e. RTCP Extended Reports, or XR).        Unfortunately, very few handsets implement these enhancements;    -   Different media streams are handled separately. Despite the fact        that they are both transmitted over the same network link, audio        and video streams are handled separately by RTCP. Both RTCP        reports provide state information about the same network,        therefore a joint analysis; and    -   Low bitrates available: The bitrate budget for a wireless        multimedia session is generally very low (under 150 kbps). The        adjustment of audio and video bitrates can have large perceptual        impact on the session, and the total available network bitrate,        even for 3G networks, can fall well below acceptable quantities.        With these issues, to wireless networks and wireless mobile        devices it has been difficult to set up a consistent streaming        media session.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a block diagram of an exemplary system.

FIG. 2 is a block diagram illustrating an embodiment of the exemplarysystem of FIG. 1.

FIG. 3 is a functional diagram illustrating an exemplary communicationflow in the exemplary system of FIG. 2.

FIG. 4 is a flowchart representing an exemplary method for processing anRTCP packet.

FIG. 5 is a flowchart representing an exemplary method for processingoptimal session bitrate data.

DETAILED DESCRIPTION OF DRAWINGS

Reference will now be made in detail to the exemplary embodimentsconsistent with the invention, the examples of which are illustrated inthe accompanying drawings. Wherever possible, the same reference numberswill be used throughout the drawings to refer to the same or like parts.

Adjusting the bitrate of streaming media sessions according toinstantaneous network capacity can be a critical function required todeliver streaming media over wireless packet networks. Adaptive bitratemanagement is a comprehensive framework and method that enables thedelivery of self-adjusting streaming sessions to media players, forexample, such as standard 3GPP-compliant media players. Adaptive bitratemanagement includes, among other things, an adaptive bitrate controllerand a variable bitrate encoder, both of which allow the adaptive bitratemanagement the ability to implement joint session bitrate management foraudio, video and/or other streams simultaneously.

FIG. 1 is a block diagram of an exemplary system. Exemplary system 100can be any type of system that transmits data packets over a network.For example, the exemplary system can include a mobile terminalaccessing streaming media data from content servers through theInternet. The exemplary system can include, among other things, aterminal 102, a gateway 104, one or more networks 106, 110, an adaptivebitrate manager 108, and one or more content servers 112-114.

Terminal 102 is a hardware component including software applicationsthat allow terminal 102 to communicate and receive packets correspondingto streaming media. Terminal 102 provides a display and one or moresoftware applications, such as a media player, for displaying streamingmedia to a user of terminal 102. Further, terminal 102 has thecapability of requesting and receiving data packets, such as datapackets of streaming media, from the Internet. For example, terminal 102can send request data to content servers 112-114 for a particular fileor object data of a web page by its URL, and the content server of theweb page can query the object data in a database and send thecorresponding response data to terminal 102. In some embodiments,response data may be routed through adaptive bitrate manager 108.

While terminal 102 can be a wired terminal, some embodiments of theinvention may prefer using a mobile terminal because mobile terminalsare more likely to be in networks that would benefit more from anadaptive bitrate manager. The network connection tends to be less stableas compared to wired network connection due to, for example, thechanging position of the mobile terminal where data rate transmissionsbetween the mobile terminal and the network can fluctuate, in some casesquite dramatically.

Gateway 104 is a device that converts formatted data provided in onetype of network to a particular format required for another type ofnetwork. Gateway 106, for example, may be a server, a router, a firewallserver, a host, or a proxy server. Gateway 104 has the ability totransform the signals received from terminal 102 into a signal thatnetwork 106 can understand and vice versa. Gateway 104 may be capable ofprocessing audio, video, and T.120 transmissions alone or in anycombination, and is capable of full duplex media translations.

Networks 106 and 110 can include any combination of wide area networks(WANs), local area networks (LANs), or wireless networks suitable forpacket-type communications, such as Internet communications. Further,networks 106 and 110 can include buffers for storing packets prior totransmitting them to their intended destination.

Adaptive bitrate manager 108 is a server that provides communicationbetween gateway 104 and content servers 112-114. Adaptive bitratemanager 108 can optimize performance by adjusting a streaming mediabitrate according to the connection, i.e., media network, betweenadaptive bitrate manager 108 and terminal 102. Adaptive bitrate manager108 can include optimization techniques, further described below.

Content servers 112-114 are servers that receive the request data fromterminal 102, process the request data accordingly, and return theresponse data back to terminal 102 through, in some embodiments,adaptive bitrate manager 108. For example, content servers 112-114 canbe a web server, an enterprise server, or any other type of server.Content servers 112-114 can be a computer or a computer programresponsible for accepting requests (e.g., HTTP, RTSP, or other protocolsthat can initiate a media session) from terminal 102 and servingterminal 102 with streaming media.

FIG. 2 is a block diagram illustrating an embodiment of the exemplarysystem of FIG. 1. Terminal 102 may include, among other things, a mediaplayer 202 and a buffer 204. Adaptive bitrate manager 108 can include,among other things, an adaptive bitrate controller 210, a buffer 212, avariable bitrate encoder 214, and a Real-time Transport Protocol (RTP)packetization 216.

Media player 202 is computer software for playing multimedia files (suchas streaming media) including video and/or audio media files. Suchpopular examples of media player 202 can include Microsoft Windows MediaPlayer, Apple Quicktime Player, and RealOne Player. In some embodiments,media player 202 decompresses the streaming video or audio using a codecand plays it back on a display of terminal 102. Media player 202 can beused as a stand alone application or embedded in a web page to create avideo application interacting with HTML content. Further, media player202 can communicate with adaptive bitrate manager 108 by sending RTCPreceiver reports.

Buffer 204 (also known as terminal buffer 204) is a software programand/or a hardware device that temporarily stores multimedia packetsbefore providing the multimedia packets to media player 202. In someembodiments, buffer 204 receives the multimedia packets from adaptivebitrate manager 108 via network 106. These packets can be configuredbased on the real-time transport protocol (RTP). In some embodiments,buffer 204 receives the multimedia packets from a device other thanadaptive bitrate manager 108. Once buffer 204 receives multimediapackets, it can provide the stored multimedia packets to media player202. While FIG. 2 illustrates that terminal buffer 204 and media player202 are separate components, one of ordinary skill the art willappreciate that terminal buffer 204 can be a part of media player 202.Further, while FIG. 2 illustrates only a single buffer, one of ordinaryskill the art will appreciate that multiple buffers can exist, forexample, one or more buffers for audio media packets and one or morebuffers for video media packets.

Adaptive bitrate controller 210 of adaptive bitrate manager 108 is asoftware program and/or hardware device that periodically receives RTCPreceiver reports from terminal 102 and provides an optimal sessionbitrate to be used during the next period for encoding multimedia datato be sent to terminal 102. In some embodiments, adaptive bitratecontroller 210 includes a buffer for storing the current and previousRTCP receiver reports. To compute the optimal session bitrate, adaptivebitrate controller 210 uses one or more network state estimators forestimating the state of the streaming media network and computing theoptimal session bitrate to be used in the next RTCP interval. Forexample, these network state estimators can estimate a media time intransit (MTT), a bitrate received at terminal 102, a round trip timeestimate (RTTE), and a packet loss count. Adaptive bitrate controller210 can use the history and statistics of the estimator to implementdifferent control algorithms to compute the optimal session bitrate.Further, adaptive bitrate controller 210 may update the optimal sessionbitrate by determining the stability of the streaming media network.This can be done by checking the newly computed estimators forcompliance to one or more stability criterion. Using the estimations andthe stability criterion, adaptive bitrate controller 210 can determinewhether to adjust the outgoing bitrate or keep the current outgoingbitrate unchanged for the next period. After this determination,adaptive bitrate controller 210 provides the optimal session bitratevalue to variable bitrate encoder 214.

Buffer 212 of adaptive bitrate manager 108 is a software program and/ora hardware device that temporarily stores media data before providingthe media data to variable bitrate encoder 214. In some embodiments,buffer 212 receives the media data from one or more content servers112-114 via network 110. In some embodiments, buffer 212 receives themedia data from a device other than content servers 112-114.

Variable bitrate encoder 214 of adaptive bitrate manager 108 is asoftware program and/or hardware device that receives optimal sessionbitrate data from adaptive bitrate controller 210 and provides, to RTPpacketization 216, audio and/or video data that are encoded at a bitratematching the optimal session bitrate provided by adaptive bitratecontroller 210. Variable bitrate encoder can include, among otherthings, a bitrate splitter 220, an audio encoder 222, a video encoder224, and, for some embodiments, a frame dropper 226.

Bitrate splitter 220 is a software program and/or a hardware device thatreceives the optimal session bitrate data from adaptive bitratecontroller 210 and allocates optimal bitrates to be used when encodingthe audio and video media data during the next interval. The allocationis such that the summation of bitrates for all tracks, when combined,can be substantially equal to the optimal session bitrate specified byadaptive bitrate controller 210. For example, this allocation could bebased on a predetermined allocation, user preference, optimalperformance data, privileging one type of data over the other, theamount of audio and video data to be provided, and/or any combination ofthe above. For example, bitrate splitter 220 may privilege audio qualityin a way that if a reduced bitrate is specified, bitrate splitter 220will reduce the video bitrate first and postpone reducing the audiobitrate as much as possible.

Audio encoder 222 and video encoder 224 are software programs and/orhardware devices that receive their respective bitrate allocation frombitrate splitter 220 and provide outgoing media data encoded to matchthe bitrate of their respective bitrate allocation for the next RTCPinterval. Both audio encoder 222 and video encoder 224 receive theirrespective media data from buffer 212 and output this media dataaccording to its respective bitrate allocation from bitrate splitter220. After the bitrate has been determined for both audio and video, itis the responsibility of each encoder to deliver maximum quality in thecorresponding media track. For example, audio encoder 222 can generatevariable bitrates by adjusting spectral quantization and cutofffrequency. Further, video encoder 224 can generate variable bitrates,for example, by adjusting Discrete Cosine Transform (DCT) coefficientquantization or by introducing frame dropping. This frame dropping canbe executed, when needed, by frame dropper 226.

Frame dropper 226 is a software program and/or a hardware device thatcan be triggered when the desired bitrate is less than a qualitythreshold. This threshold can be codec dependent, and represents thebitrate value below which the use of coarser quantization leads tointolerable artifacts in the image. Frame dropper 226 can dynamicallydetermine a frame dropping rate based on the desired video bitrate andthe bitrate being generated by video encoder 224. To compensate inherentbitrate fluctuations in the video bitrate at the output of the encoder,frame dropper 226 can dynamically update the dropping rate by using asliding window covering the byte size history of recently encodedframes.

RTP packetization 216 is a software program and/or a hardware devicethat receives the audio and video media data from audio encoder 222 andvideo encoder 224 and translates this data into a packet format. RTPdefines a standardized packet format for delivering audio and video overthe Internet. Besides carrying the audio and media data, these packetscan include, among other things, a payload-type identifier foridentifying the type of content, a packet sequence number, time stampingfor allowing synchronization and jitter calculations, and deliverymonitoring data. This type of data can later assist adaptive bitratecontroller 210 in determining the quality of service provided by thenetwork when adaptive bitrate controller 210 receives a correspondingRTCP receiver report from terminal 102. Upon translating this data intoa packet format, RTP packetization 216 transmits the data throughnetwork buffer 230 of network 106 to terminal buffer 204 of terminal102. In addition adaptive bitrate manager 108 saves the history of sentRTP packets in the audio and video tracks. This history data caninclude, among other things, the time that each packet is sent, thesequence number, and the size of each RTP packet.

FIG. 3 is a functional diagram illustrating an exemplary communicationflow in the system of FIG. 2. It is assumed for purposes of explainingthis exemplary embodiment that terminal 102 has already received atleast some of the media data of the requested media data package.Further, it is assumed that the media data package includes both audioand video media data. After receiving packets, media player 202transmits (302) an RTCP receiver report to adaptive bitrate manager 108.

RTCP is a protocol for providing quality control information for an RTPflow, such as the transmission provided by RTP packetization 216 ofadaptive bitrate manager 108. More specifically, RTCP partners with RTPpacketization 216 of adaptive bitrate manager 108 in the delivery andpackaging of multimedia data. In some embodiments, media player 202periodically transmits the RTCP receiver report. RTCP receiver reportcan provide feedback on the quality of service being provided by RTPpacketization 216. While RTP/RTCP is used as an exemplary embodiment toexplain the adaptive bitrate control method, one of ordinary skill couldappreciate that this adaptive bitrate control method is applicable toany protocol that fulfills the functions of media transport withsequencing and timing information and media transport feedback withinformation about received packets (covering sequencing, timing, lossrate, etc.)

Further, in some embodiments, the receiver report can be a single reporthaving both audio and video report data or it can be separated intomultiple reports (such as in the RTCP case), for example, such as areceiver report for audio report data and another receiver report forvideo report data. The receiver report data can include, among otherthings, data regarding the sequence number of the most recently receivedRTP packet at terminal 102, the timestamp of the last packet received byterminal 102 reported in the RTCP receiver report, the number of bitssent from this report, a round trip time, and a number of packets lost.

After receiving the receiver report, adaptive bitrate controller 210 canestimate the state of the network for determining whether to update thesession bitrate for the next period. Adaptive bitrate controller 210 cansave the newly received receiver report in a cumulative history andrecord the time at which the packet was received. To estimate the stateof the network, adaptive bitrate controller 210 can combine data fromthe received RTCP receiver report, the previously received RTCP receiverreports stored by the adaptive bitrate manager 108, and the history ofsent RTP packets stored by adaptive bitrate manager 108. Adaptivebitrate controller can estimate the following exemplary data by usingnetwork state estimators:

-   -   Media Time in Transit (MTT), computed as the difference between        the timestamp of the most recently sent RTP packet and the        timestamp of the last RTP packet received by the player reported        in RTCP receiver report;    -   Bitrate received, computed as the bits received between the        current and previously received RTCP receiver reports, divided        by the time elapsed between these two receiver reports. The bits        received between receiver reports are computed by cross        referencing sequence numbers in the receiver report with the        history of bytes sent stored at adaptive bitrate manager 108;    -   Round Trip Time Estimate (RTTE) can be obtained by averaging a        number of the lower MTT values stored at the adaptive bitrate        manager 108. For example, RTTE could be calculated by averaging        the lowest 3 MTT values out of all stored MTT values for that        streaming media network. Further, adaptive bitrate manager 108        can calculate the RTTE from data within an RTCP sender report.        While these exemplary embodiments are illustrated, any method        can be used to estimate a round trip time for the streaming        media network; and    -   Packet Loss count, captured directly from RTCP receiver report.        Adaptive bitrate controller 210 can use these estimates to        implement several different control algorithms. For example, the        Streaming Media stability criterion can be used to compute the        session bitrate for the next RTCP interval.

Adaptive bitrate controller 210 uses the stability criterion todetermine the stability of the streaming media network. While any numberof algorithms can be used to determine the stability, one exemplaryembodiment compares the estimated MTT with the RTTE. If the MTT and theRTTE remain close, adaptive bitrate controller 210 can determine thatthe streaming media network can properly support the current bitrate.Further, by comparing the bitrate received with the current bitratesession, adaptive bitrate controller 210 can determine that the networkcan cope with the load imposed by adaptive bitrate manager 108.

Adaptive bitrate controller 210 uses the estimations and the stabilitycriterion to implement control algorithms for discovering the networkcapacity and adjusting the session bitrate accordingly. Adaptive bitratecontroller 210 can define the variations of the control algorithms tooperate in two different modes: (1) acquisition mode and (2) normalmode. While two modes have been illustrated in this exemplaryembodiment, one of ordinary skill in the art will appreciate thatmultiple modes of operation can be defined.

In the normal mode, adaptive bitrate controller 210 operates in thesteady state condition, indicating that the network is eithermaintaining or incrementally increasing the effective capacity seen bythe system. In some embodiments, while operating in normal mode, thecontrol algorithms can increase the session bitrate while the MTT is notincreasing and the bitrate received remains close to the current sessionbitrate.

Adaptive bitrate controller 210 generally triggers the acquisition modewhen it detects high packet loss, a sudden increase in the MTT, and/or avalue of the MTT higher than a threshold (MTT threshold), which can be afixed value or can be obtained dynamically for an adaptive controlmechanism. Once triggered, acquisition mode sets the optimal sessionbitrate to a value, such as the bitrate received or a fraction of thereceived bitrate. Because the bitrate received can be the bestestimation of the actual bitrate that the network can support at thatparticular point in time, adaptive bitrate manager 108 should quicklyreturn back to a stable condition. In some embodiments, the new sessionbitrate is simply set to be a fraction of the current session bitrate.

In this embodiment, while only terminal 102 is illustrated forcommunicating with adaptive bitrate manager 108, one of ordinary skillin the art will appreciate that multiple terminals can communicate withadaptive bitrate manager 108, where each of the terminals can be locatedin substantially different network environments. Such environments canvary significantly, as different underlying wireless technologies andfixed network topologies can be used. Therefore, for some embodiments,it may be desirable to discover characteristics of the networkenvironment beforehand so that key parameters in the framework areadjusted automatically. For example, adaptive bitrate controller 210could set the MTT threshold at the beginning of the multimedia sessionto a value correlated to the RTTE. In this way, the system can attemptto follow the general stability criterion provided by adaptive bitratecontroller 210. As indicated above, this stability criterion could bebased on, independent of the network environment (a prior unknown), thecomparison between the MTT and the RTTE, which is largely advantageousgiven that the actual network infrastructure type can rarely bedetermined a priori. In some embodiments, the optimal session bitratecan be updated by determining the difference between the MTT and theRTTE and adjusting the session bitrate according to the difference. Forexample, the larger the difference, the greater adjustment from thecurrent session bitrate to an optimal session bitrate. In someembodiments, the MTT used for this determination can be based on the oneor more historical values of MTT.

Using the control algorithms to compute a session bitrate update asdescribed above, adaptive bitrate controller 210 determines an optimalsession bitrate for transmitting media data to terminal 102. Adaptivebitrate controller 210 provides (304) the optimal session bitrate datato bitrate splitter 220 of variable bitrate encoder 214. Upon receivingthe optimal session bitrate data, bitrate splitter 220 allocates theoptimal session bitrate between the audio and video streams. Forexample, this allocation could be based on a predetermined allocation, auser preference optimal performance data, privileging one type of dataover the other, the amount of audio and video data to be provided,and/or any combination of the above. For example, bitrate splitter 220may privilege audio quality in a way that if a reduced bitrate isspecified, bitrate splitter 220 reduces the video bitrate first andpostpones reducing the audio bitrate as much as possible.

After splitting the optimal session bitrate into an optimal audiobitrate and an optimal video bitrate, bitrate splitter provides (306)the optimal audio bitrate to audio encoder 222 and provides (308) theoptimal video bitrate to video encoder 224. Upon receiving theirrespective bitrate, both audio encoder 222 and video encoder 224 receivetheir respective media data from buffer 212 and output their respectiveaudio media data and video media data according to the respectivebitrate allocation from bitrate splitter 220. After the bitrate has beendetermined for both audio and video, it is the responsibility of eachencoder to deliver maximum quality in the corresponding media track bymaintaining the requested bitrate until the next RTCP interval. Forexample, audio encoder 222 can generate variable bitrates by adjustingquantization and cutoff frequency. Further, video encoder 224 cangenerate variable bitrates, for example, by adjusting Discrete CosineTransform (DCT) coefficient quantization or by introducing framedropping. This frame dropping can be executed, when needed, by framedropper 226. In some embodiments, the encoding parameters of theencoders are not modified until they receive optimal bitrate data frombitrate splitter 220, which would be provided in a subsequent RTCPinterval, because the encoders 222, 224 are slave devices to bitratesplitter 220.

In some embodiments, where frame dropping is preferred, video encoder224 can provide (310) the video media data to frame dropper 226 when theoptimal session bitrate is less than a quality threshold. This thresholdcan be codec dependent, and represents the bitrate value below which theuse of coarser quantization leads to intolerable artifacts in the image.When frame dropping is triggered, frame dropper 226 can dynamicallydetermine a frame dropping rate based on the desired video bitrate andthe bitrate being generated by video encoder 224. To compensate inherentbitrate fluctuations in the video bitrate at the output of video encoder224, frame dropper 226 can dynamically update the dropping rate by usinga sliding window covering the byte size history of recently encodedframes. Frame dropper 226 can drop the frames accordingly to deliver theoptimal session bitrate. In addition, in some embodiments, video encoder224 can utilize the network state estimator of adaptive bitratecontroller 210 to encode video in a more resilient manner. In someembodiments, packet loss information can be used in conjunction with theMTT by video encoder 224 to determine if a Group of Picture (GOP) valueshould be reduced, increasing the number of I-Frames per second sent inthe video stream. In some embodiment, if frame dropping is not needed,video encoder 224 can simply provide the video media data to RTPpacketization 216. Audio encoder 222 and, for this embodiment, framedropper 226 provide (312, 314) the audio media data and the video mediadata, respectively, to RTP packetization 216.

Upon receiving the audio media data and the video media data, RTPpacketization 216 translates this data into a packet format. RTP definesa standardized packet format for delivering audio and video over theInternet. Upon translating this data into a packet format, RTPpacketization 216 transmits (316) the audio and video media packets tonetwork buffer 230 of network 106. While only one transmission is shown,one of ordinary skill in the art will appreciate that transmission 316can include separate transmission for one or more audio media packetsand another for one or more video media packets. Furthermore, one ofordinary skill in the art will appreciate that network 106 can includemultiple networks, each having their own one or more buffers. Besidescarrying the audio and media data, these packets can include, amongother things, a payload-type identifier, a packet sequence number, atimestamp, and delivery monitoring data. This type of data can laterassist adaptive bitrate controller 210 in determining the quality ofservice provided by the network when adaptive bitrate controller 210receives the RTCP receiver report from terminal 102. Moreover, adaptivebitrate manager 108 can also store a history of sent RTP packets so thatit can later adjust the bitrate accordingly.

Upon receiving the packets, network buffer 230 of network 106 can storethe packets until it is the packets turn to be provided to terminal 102.While only buffer 230 is illustrated, one of ordinary skill in the artwill appreciate that one or more separate buffers can exist for each ofthe audio media packets and the video media packets. When it is thepackets turn, network buffer 230 transmits (318) the packets to terminalbuffer 204.

Upon receiving the packets, terminal buffer 204 of terminal 102 canstore the packets until it is the packets turn to be provided to mediaplayer 202. While only buffer 230 is illustrated, one of ordinary skillin the art will appreciate that one or more separate buffers can existfor each of the audio media packets and the video media packets. When itis the packets turn, buffer 204 provides (320) the packets to mediaplayer 202. In turn, media player 202 can extract the relevant data outof packets and provide this data to adaptive bitrate manager 108 in asubsequent RTCP receiver report.

FIG. 4 is a flowchart representing an exemplary method for processing anRTCP packet. Referring to FIG. 4, it will be readily appreciated by oneof ordinary skill in the art that the illustrated procedure can bealtered to delete steps or further include additional steps. It isassumed for this exemplary method that RTCP packet includes dataconcerning both audio and video media data. While both types exists, oneof ordinary skill in the art will appreciate that RTCP data can includeeither audio or video data. After initial start step 400, an adaptivebitrate manager obtains (402) RTCP data, which can include one or moreRTCP receiver reports. This RTCP data can correlate to the quality andquantity of audio and video media packets received at a media player ofa terminal. The RTCP data can include, among other things, a sequencenumber of a last packet received by the terminal, a timestampcorresponding to such packet, a number of bits sent, a round trip time,and number of packets lost during a transmission from the adaptivebitrate manager to the terminal. RTCP data can be obtained by receivingan RTCP report from the terminal and by cross-correlating the contentsof the last received RTCP packet with the history of RTP packets storedat the adaptive bitrate manager.

After receiving RTCP data, the adaptive bitrate manager estimates (404)network conditions of a streaming media network. To estimate the stateof the network, the adaptive bitrate manager can combine data from thereceived RTCP data from step 402 and previously received RTCP datastored by the adaptive bitrate manager. Adaptive bitrate controller canestimate an MTT, a bitrate received, an RTTE, and a packet loss. Theadaptive bitrate manager can use these estimates to implement severaldifferent control algorithms.

After estimating the network conditions, the adaptive bitrate managerapplies (406) stability criterion to determine the stability of thestreaming media network. If needed, the stability criterion can assistin adjusting the bitrate for attempting to stabilize the streaming medianetwork, e.g., such as avoiding buffer overflows in the network andunderflows at the terminal. While any number of algorithms can be usedto determine the stability criterion, one exemplary embodiment comparesthe estimated MTT with the estimated RTTE, both of which are estimatedin step 404. If the MTT and the RTTE remain close, the adaptive bitratemanager can use this comparison to determine that the streaming medianetwork can properly support the current bitrate. Further, by comparingthe bitrate received with the current bitrate session, the adaptivebitrate manager can determine that the streaming media network can copewith the load.

After establishing the stability criterion, the adaptive bitrate managerdetermines (408) whether the network is stable with respect to thecurrent bitstream based on estimation step 404 and/or stabilitycriterion establishment step 406. If the network is stable, the adaptivebitrate manager operates (410) in a steady state condition by eithermaintaining or incrementally increasing the current bitrate. In someembodiments, the optimal session bitrate can be computed by determiningthe difference between the MTT and the RTTE and adjusting the sessionbitrate according to the difference. For example, if the current sessionbitrate is less than a set target session bitrate, the adaptive bitratemanager can incrementally increase the optimal session bitrate if thevalues of the MTT and the RTTE are comparable. Then, the adaptivebitrate manager provides (416) an optimal session bitrate fortransmitting media data to a terminal. After providing step 416, themethod can proceed to end 418.

If determining that the network is not stable, the adaptive bitratemanager adjusts (412) the bitrate so that adaptive bitrate manager canreach a stable condition. For example, in some embodiments, the adaptivebitrate manager can use the estimated bitrate received from step 404because, in some embodiments, the bitrate received can be the bestestimation of the actual bitrate that the network can support at thatparticular point in time. Then, the adaptive bitrate manager provides(416) the optimal session bitrate for transmitting media data to theterminal. After providing step 416, the method can proceed to end 418.

FIG. 5 is a flowchart representing an exemplary method for processingoptimal session bitrate data. Referring to FIG. 5, it will be readilyappreciated by one of ordinary skill in the art that the illustratedprocedure can be altered to delete steps or further include additionalsteps. It is assumed for this exemplary method that both audio and videomedia data exists. While both types exists, one of ordinary skill in theart will appreciate that either audio or video data can exist. Afterinitial start step 500, an adaptive bitrate manager obtains (502)optimal session bitrate data for transmitting media data to a terminal.

Upon receiving the optimal session bitrate data, the adaptive bitratemanager allocates (504) the optimal session bitrate between audio andvideo streams to produce an optimal audio bitrate and an optimal videobitrate. For example, this allocation could be based on a predeterminedallocation, user preference, optimal performance data, privileging onetype of data over the other, the amount of audio and video data to beprovided, and/or any combination of the above. For example, the adaptivebitrate manager may privilege audio quality in a way that if a reducedbitrate is specified, the adaptive bitrate manager can reduce the videobitrate first and postpone reducing the audio bitrate as much aspossible.

Adaptive bitrate manager obtains (506) audio and video media data. Insome embodiments, obtaining step 506 can occur prior to allocating step504 or obtaining step 502. After allocating step 504 and obtaining step506, the adaptive bitrate manager encodes (508) the audio and videomedia data according to their respective allocated bitrate specified atstep 504.

After encoding the audio and video streams according to the allocatedbitrate, the adaptive bitrate manager provides (510) the encoded audioand video media data for transmitting to the terminal. In someembodiments, an RTP packetization receives the encoded audio and videomedia data and translates this data into a packet format. RTP defines astandardized packet format for delivering audio and video over theInternet. Upon translating this data into a packet format, the RTPpacketization can then transmit the audio and video media packets to theterminal. After providing the encoded audio and video media data, themethod can proceed to end 512.

The methods disclosed herein may be implemented as a computer programproduct, i.e., a computer program tangibly embodied in an informationcarrier, e.g., in a machine readable storage device or in a propagatedsignal, for execution by, or to control the operation of, dataprocessing apparatus, e.g., a programmable processor, a computer, ormultiple computers. A computer program can be written in any form ofprogramming language, including compiled or interpreted languages, andit can be deployed in any form, including as a stand alone program or asa module, component, subroutine, or other unit suitable for use in acomputing environment. A computer program can be deployed to be executedon one computer or on multiple computers at one site or distributedacross multiple sites and interconnected by a communication network.

In the preceding specification, the invention has been described withreference to specific exemplary embodiments. It will however, be evidentthat various modifications and changes may be made without departingfrom the broader spirit and scope of the invention as set forth in theclaims that follow. The specification and drawings are accordingly to beregarded as illustrative rather than restrictive. Other embodiments ofthe invention may be apparent to those skilled in the art fromconsideration of the specification and practice of the inventiondisclosed herein.

1. A method comprising: receiving a receiver report from a terminal having a media player; estimating one or more network conditions of a media network using the receiver report; determining an optimal session bitrate using the estimated one or more network conditions; allocating the optimal session bitrate between audio media data and video media data to produce an optimal audio bitrate and an optimal video bitrate, wherein allocating the optimal session bitrate between audio and video media data is based on a metric selected from a group including a predetermined allocation, a user preference, an optimal performance data, privileging one type of data over the other, and an amount of audio and video media data to be provided; encoding audio and video media data according to the optimal audio bitrate and the optimal video bitrate; and providing media data to the terminal according to the optimal audio bitrate and the optimal video bitrate.
 2. The method of claim 1, wherein determining the optimal session bitrate further comprises: establishing stability criterion based on the estimated one or more network conditions; determining the stability of the media network; and providing the optimal session bitrate based on the stability determination.
 3. The method of claim 2, wherein establishing stability criterion further comprises comparing a media time in transit and a round trip time estimate for assisting with the stability determination.
 4. The method of claim 2, wherein establishing stability criterion further comprises comparing a bitrate received with a current bitrate session.
 5. The method of claim 2, further comprising maintaining or incrementally increasing a current bitrate when the stability of the media network is considered normal.
 6. The method of claim 2, further comprising adjusting a current bitrate when the stability of the network is not normal.
 7. The method of claim 1, wherein providing media data to the terminal includes providing the encoded audio media data and the encoded video media data based on the optimal audio bitrate and the optimal video bitrate.
 8. The method of claim 1, wherein a receiver report is selected from a group including a real-time transport control protocol (RTCP) receiver report, and a real-time transport control protocol extended reports (RTCP XR) receiver report.
 9. A method comprising: receiving a receiver report from a terminal; estimating one or more network conditions of a media network using the receiver report; determining the stability of the media network using the estimations; controlling a bitrate based on the determination; allocating the bitrate between audio media data and video media data to produce an optimal audio bitrate and an optimal video bitrate, wherein allocating the bitrate between audio and video media data is based on a metric selected from a group including a predetermined allocation, a user preference, an optimal performance data, privileging one type of data over the other, and an amount of audio and video media data to be provided; and providing the bitrate to an encoder for encoding audio and video media data, and transmitting media data according to the optimal audio bitrate and the optimal video bitrate.
 10. The method of claim 9, wherein determining the stability of the media network is based on establishing stability criterion.
 11. The method of claim 10, wherein establishing stability criterion further comprises comparing a media time in transit and a round trip time estimate for assisting with the stability determination.
 12. The method of claim 10, wherein establishing stability criterion further comprises comparing a bitrate received with a current bitrate session.
 13. The method of claim 9, wherein controlling the bitrate includes maintaining or incrementally increasing a current bitrate when the stability of the media network is considered normal.
 14. The method of claim 9, wherein controlling the bitrate includes adjusting a current bitrate when the stability of the network is considered not normal.
 15. The method of claim 9, wherein a receiver report is selected from a group including a real-time transport control protocol (RTCP) receiver report, and a real-time transport control protocol extended reports (RTCP XR) receiver report.
 16. A method comprising: receiving an optimal session bitrate; allocating the optimal session bitrate between audio and video media data to produce an optimal audio bitrate and an optimal video bitrate, wherein allocating the optimal session bitrate between audio and video media data is based on a metric selected from a group including a predetermined allocation, a user preference, an optimal performance data, privileging one type of data over the other, and an amount of audio and video media data to be provided; encoding audio and video media data according to the optimal audio bitrate and the optimal video bitrate; and providing the encoded audio and video data for transmittal to a terminal.
 17. The method of claim 16, further comprising dropping frames of the encoded video data.
 18. A system comprising: an adaptive bitrate manager comprising a processor and a memory, the adaptive bitrate manager configured to: receive a receiver report provided by a terminal having a media player, estimate one or more network conditions using the receiver report, determine an optimal session bitrate based on the estimated one or more network conditions, and provide media data to the terminal according to the optimal session bitrate, and wherein the adaptive bitrate manager further comprises an encoder configured to: obtain the optimal session bitrate, allocate the optimal session bitrate between audio and video media data to produce an optimal audio bitrate and an optimal video bitrate, wherein allocating the optimal session bitrate between audio and video media data is based on a metric selected from a group including a predetermined allocation, a user preference, an optimal performance data, privileging one type of data over the other, and an amount of audio and video media data to be provided, and encode audio and video media data according to the optimal audio bitrate and the optimal video bitrate, and provide the encoded audio and video media data for transmittal to the terminal.
 19. The system of claim 18, wherein the adaptive bitrate manager further comprises an adaptive bitrate controller configured to receive the receiver report and calculating an optimal session bitrate.
 20. The system of claim 18, wherein a receiver report is selected from a group including a real-time transport control protocol (RTCP) receiver report, and a real-time transport control protocol extended reports (RTCP XR) receiver report.
 21. A non-transitory computer readable storage medium storing instruction that, when executed by a computer, cause the computer to perform a method for processing a receiver report, the method comprising: receiving the receiver report from a terminal; estimating one or more network conditions of a media network using the receiver report; determining the stability of the media network using the estimations; controlling a bitrate based on the determination; allocating the bitrate between audio media data and video media data to produce an optimal audio bitrate and an optimal video bitrate, wherein allocating the bitrate between audio and video media data is based on a metric selected from a group including a predetermined allocation, a user preference, an optimal performance data, privileging one type of data over the other, and an amount of audio and video media data to be provided; and providing the bitrate to an encoder for transmitting media data according to the optimal audio bitrate and the optimal video bitrate.
 22. The computer readable medium of claim 21, wherein a receiver report is selected from a group including a real-time transport control protocol (RTCP) receiver report, and a real-time transport control protocol extended reports (RTCP XR) receiver report.
 23. A non-transitory computer readable storage medium storing instruction that, when executed by a computer, cause the computer to perform a method for processing an optimal session bitrate, the method comprising: receiving the optimal session bitrate; allocating the optimal session bitrate between audio and video media data to produce an optimal audio bitrate and an optimal video bitrate, wherein allocating the optimal session bitrate between audio and video media data is based on a metric selected from a group including a predetermined allocation, a user preference, an optimal performance data, privileging one type of data over the other, and an amount of audio and video media data to be provided; encoding audio and video media data according to the optimal audio bitrate and the optimal video bitrate; and providing the encoded audio and video media data for transmittal to a terminal.
 24. A terminal comprising: a processor; a memory; a buffer configured to receive media data packets transmitted by an adaptive bitrate manager over a media network; and a media player configured to receive media data packets and provide a receiver report to the adaptive bitrate manager configured to: receive the receiver report, estimate one or more network conditions of the media network based at least in part on the receiver report, determine an optimal session bitrate based on the estimated one or more network conditions, and provide media data to the buffer according to the optimal session bitrate, wherein the adaptive bitrate manager further comprises an encoder configured to: obtain the optimal session bitrate, allocate the optimal session bitrate between audio and video media data to produce an optimal audio bitrate and an optimal video bitrate, wherein allocating the optimal session bitrate between audio and video media data is based on a metric selected from a group including a predetermined allocation, a user preference, an optimal performance data, privileging one type of data over the other, and an amount of audio and video media data to be provided encode audio and video media data according to the optimal audio bitrate and the optimal video bitrate, and provide the encoded audio and video media data for transmittal to the terminal.
 25. The terminal of claim 24, wherein a receiver report is selected from a group including a real-time transport control protocol (RTCP) receiver report, and a real-time transport control protocol extended reports (RTCP XR) receiver report. 